SIP Video

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2007 Avaya Inc. All rights reserved.

SIP Videowith Communication Manager

Chris Kendall

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2007 Avaya Inc. All rights reserved.Avaya Proprietary

The addition of SIP to Avayas Video Telephony Solution provides theframework for the future of enterprise video conferencing

Leveraging existing IETF open standards

Ensuring higher levels of interoperability with third parties

Maintaining a sophisticated feature set that is consistent and

compatible with todays H.323 video solutionHighlights include:

H.323-SIP inter-working, with a common subset of telephony features

Call admission control and video enablement policies

Bandwidth management

Priority video callers Cumulative pools for audio and video

Call rate negotiation

Video Conferencing (scheduled and ad-hoc)

Overview Description

SIP Video

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Solution view

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Solution Components

CM is the

Feature server - extends many CM features to SIP phonesthru OPTIM architecture

Gatewayinterworks SIP to all other protocols (H.323, ISDN,DCP, analog, etc..)

Back-to-back user-agent (B2BUA)

SES contains the following components

Proxy

Registrar Event Server

Personal Profile Manager (PPM)

Location service (database)

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Solution Components (continued)Supported SIP Adjuncts

Modular Messaging

Voice Portal

Meeting Exchange (version 5 with Video)

Supported SIP Endpoints

Avaya SIP Softphone (AST)

46xx SIP

96xx SPARK phone (AST)

Toshiba SIP phone (AST)

One-X mobile edition (CHAMP)

Cisco and other third-party SIP phones

SIP Video phones NEW!

NEW!

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Meeting Exchange

Meeting Exchange is the existing large scale SIP conferencing platform

for CM

Version 5 supports video conferencing of up to 16 simultaneoususers.

Meet-me, scheduled and ad-hoc conferences may be used

Configured as a video-enabled trunk on CM Video-bridge configuration must be set for ad-hoc conferencing

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7/19 2007 Avaya Inc. All rights reserved.

Avaya Proprietary

SIP Video Phones

Most Polycom & Tandberg room systems are dual-protocol H.323 and

SIP Tandberg T150 (v4.1) has been tested extensively with CM in SIP

For presentations and collaboration, H.323 is still recommended

There are many third party SIP phones out in the market, especiallySoftphones that can be connected to SES

Some will work as OPTIM endpoints, i.e. CM-routed and otherswill not

Avayas next generation SIP Softphone (with video) is TBA

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Avaya Proprietary

Jargon Buster

Lots of jargon and acronyms to digest SIP: Session Initiation Protocol

SDP: Session Description Protocol

Dialog: A conversation/communication

Session: Media (audio/video/data) between two entities

AST: Advanced SIP Telephony

SUSHI: Toshiba SIP Phone (supporting AST)

B2BUA: Back-to-Back User Agent

OPTIM/OPS: Off PBX/Premises Station

OATS: Origination And Termination Service

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Avaya Proprietary

High-level Call-flow

Edge ProxySIP phone-A

NJ CM

SIP phone-B

DR Home Proxy NJ Home Proxy

DR CM

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Avaya Proprietary

Why OPTIM?

Using OPTIM architecture, CM becomes a central point of control forvideo policy

BWM, CAC, video and conferencing capabilities

As a B2BUA, CM will modify the SDP to enforce these policies

Allow or deny video

Restrict call-ratesAllow/deny use of shared resources

As a SIP-H.323 gateway, CM bridges the protocol gap between differentusers and systems

Dont need to dial special addresses

Dont need to think about protocols or devices

Telephony features just work

Video is still as easy as making a phone call

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11/19 2007 Avaya Inc. All rights reserved.Avaya Proprietary

SIP H.323 Interoperability

CM places particular requirements on endpoints for H.323

interoperability:

RFC3890, for enterprise bandwidth management

SIP endpoints to present their full capabilities in the SDP

Supporting asymmetric SDP payload types is also required

Some SIP endpoints can be configured to support the functionalityrequired by CM

For example, In Counterpath Eyebeam (1.5) you can dial ***7469 toreveal a hidden config UI and set the following options:

Rtp:media:send_bandwidth_modifier = 1

Media:sdp:specify_all_codecs_in_offer_answer = 1

Media:sdp:force_describe_well_known_codecs = 1

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Solution Limitations

This release has some limitations with CM as the inter-workingSIP-H.323 gateway using video endpoints.

SIP video endpoints are not supported dialing into thePolycom MGC (either H.323 or SIP)

SIP video endpoints cannot view H.239 streams (e.g.Polycom People plus Content)

Polycom SIP firmware is not currently supported whenconfigured as an OPTIM endpoint

Multipoint SIP room systems are not supported with CM

Future releases aim to resolve these limitations

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SIP Priority Video Users

Priority video is not signaled over SIP trunks

So it works a bit differently

If Priority video is enabled and the SIP signaling group is terminating thecall:

The priority status of the originating caller is preserved

If Priority video is enabled and there is an incoming call on the SIPsignaling group:

The call gets promoted to priority

SIP sub-domains may therefore be used to separate priority users from

regular users [email protected]

[email protected]
mailto:[email protected]:[email protected]:[email protected]:[email protected]

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SES Administration

SES Administration is effectively unchanged from existingSIP audio deployments.

Primary consideration is whether a user is mapped to amedia server extension (CM station) or not

Video calls are unmanaged when they are pure SIP anddo not route via CM.

Calling a H.323 user will still route the call through CM if themedia-server address maps are configured

This also holds true for calls involving OPTIM SIP users

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CM Admin Licensing

New Feature to be enabled in license file.

Multimedia IP SIP Trunking? (FEAT_MMIP_SIP)

Multimedia IP SIP Trunking? can then be enabled in customer options

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CM Admin SIP Video Trunk

Two new fields for video on the SIP signaling group.

IP Video and Priority Video

SIP trunk-group administration is unchanged

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CM Admin OPTIM stations

New set type 4620SIP should be used for all non-AST SIP endpoints

IP Video should be enabled, while IP Softphone should be disabled

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Common Problems

The call fails to connect when calling from a SIP endpoint If response is 403 forbidden (Denial 11), check COR/COS, ip-network-region domain

If response is 482 Loop detected, check dial plan (UDP/AAR/EXT)

If response is 488 Not acceptable, check audio codec is supported

I get no video when calling from a SIP endpoint Check licensing and capacities

Check signaling group, network-region and codec-set admin

Check station admin - ensure IP-direct audio enabled for station form page 2

Check bandwidth is available

Check endpoint enabled for video, and is offering bandwidth

Due to protocol complexity, SIP users will not get video dialing into the H.323 MGC via CM

My SIP endpoint wont register If response is 401 Unauthorized, check password

If response is 404 User not found, check username and domain are correct,

check that the SES configuration has been saved/updated

More items in the AVTS configuration checklist

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References and Resources

SIP support in Communication Manager 4.0:

http://support.avaya.com/elmodocs2/sip/245206_7.pdf

Installing and administering SES:

http://support.avaya.com/elmodocs2/sip/03_600768_4.pdf
http://support.avaya.com/elmodocs2/sip/245206_7.pdfhttp://support.avaya.com/elmodocs2/sip/03_600768_4.pdfhttp://support.avaya.com/elmodocs2/sip/03_600768_4.pdfhttp://support.avaya.com/elmodocs2/sip/245206_7.pdf

Documents

SIP Video